http://blog.csdn.net/sepnic/article/details/7307506
之前轉載過一篇文章-智能手機音頻系統概述,描述了手機音頻系統設計框圖。實際上那是一個簡單的做法,應用中有較大的侷限性。那麼一個完善的音頻框架應該是什麼樣的呢?這兩天根據Android4.0源碼的一些線索,找到了相應的硬件資料,摘錄下來。
注:以samsung tuna方案(即galaxy nexus)爲例。
audio_hw
在ANDROID音頻系統散記之四:4.0音頻系統HAL初探中,提及到samsung的tuna方案,其實就是大名鼎鼎的galaxy nexus了。
android-4.0.3_r1\device\samsung\tuna\audio\audio_hw.c,這文件就是tuna的音頻HAL了,從中我們看出:根據上層的音頻策略打開/關閉相應的pcm設備。
- struct pcm_config pcm_config_mm = {
- .channels = 2,
- .rate = MM_FULL_POWER_SAMPLING_RATE,
- .period_size = LONG_PERIOD_SIZE,
- .period_count = PLAYBACK_LONG_PERIOD_COUNT,
- .format = PCM_FORMAT_S16_LE,
- };
- struct pcm_config pcm_config_mm_ul = {
- .channels = 2,
- .rate = MM_FULL_POWER_SAMPLING_RATE,
- .period_size = SHORT_PERIOD_SIZE,
- .period_count = CAPTURE_PERIOD_COUNT,
- .format = PCM_FORMAT_S16_LE,
- };
- struct pcm_config pcm_config_vx = {
- .channels = 2,
- .rate = VX_NB_SAMPLING_RATE,
- .period_size = 160,
- .period_count = 2,
- .format = PCM_FORMAT_S16_LE,
- };
struct pcm_config pcm_config_mm = {
.channels = 2,
.rate = MM_FULL_POWER_SAMPLING_RATE,
.period_size = LONG_PERIOD_SIZE,
.period_count = PLAYBACK_LONG_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
struct pcm_config pcm_config_mm_ul = {
.channels = 2,
.rate = MM_FULL_POWER_SAMPLING_RATE,
.period_size = SHORT_PERIOD_SIZE,
.period_count = CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
struct pcm_config pcm_config_vx = {
.channels = 2,
.rate = VX_NB_SAMPLING_RATE,
.period_size = 160,
.period_count = 2,
.format = PCM_FORMAT_S16_LE,
};
1、mm:media playback設備,即audio download link;
2、mm_ul:audio record設備,即audio upload link;
3、vx:voice設備,通話模塊的聲音就是經過這個設備的。
根據上層聲音模式audio_mode_t來選擇打開不同的pcm設備,詳細見select_mode()函數。
audio_hw.c還定義了各種音頻路徑(音頻路徑概念見:DAPM之二:audio paths與dapm kcontrol)。
- struct route_setting hf_output[] = {
- {
- .ctl_name = MIXER_HF_LEFT_PLAYBACK,
- .strval = MIXER_PLAYBACK_HF_DAC,
- },
- {
- .ctl_name = MIXER_HF_RIGHT_PLAYBACK,
- .strval = MIXER_PLAYBACK_HF_DAC,
- },
- {
- .ctl_name = NULL,
- },
- };
- struct route_setting hs_output[] = {
- {
- .ctl_name = MIXER_HS_LEFT_PLAYBACK,
- .strval = MIXER_PLAYBACK_HS_DAC,
- },
- {
- .ctl_name = MIXER_HS_RIGHT_PLAYBACK,
- .strval = MIXER_PLAYBACK_HS_DAC,
- },
- {
- .ctl_name = NULL,
- },
- };
- // ......
struct route_setting hf_output[] = {
{
.ctl_name = MIXER_HF_LEFT_PLAYBACK,
.strval = MIXER_PLAYBACK_HF_DAC,
},
{
.ctl_name = MIXER_HF_RIGHT_PLAYBACK,
.strval = MIXER_PLAYBACK_HF_DAC,
},
{
.ctl_name = NULL,
},
};
struct route_setting hs_output[] = {
{
.ctl_name = MIXER_HS_LEFT_PLAYBACK,
.strval = MIXER_PLAYBACK_HS_DAC,
},
{
.ctl_name = MIXER_HS_RIGHT_PLAYBACK,
.strval = MIXER_PLAYBACK_HS_DAC,
},
{
.ctl_name = NULL,
},
};
// ......
1、hf_output:headfree輸出路徑;
2、hs_output:headset輸出路徑;
3、......
根據上層的audio_devices_t選擇打開或關閉對應的音頻路徑部件,如:
- // ...
- headset_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- headphone_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- // ...
- set_route_by_array(adev->mixer, hs_output, headset_on | headphone_on);
- set_route_by_array(adev->mixer, hf_output, speaker_on);
- // ...
// ...
headset_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
headphone_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
// ...
set_route_by_array(adev->mixer, hs_output, headset_on | headphone_on);
set_route_by_array(adev->mixer, hf_output, speaker_on);
// ...
另外有一個我非常疑惑的地方:tuna的audio_hw.c中選用的採樣率是48khz,但是Android的framework層是先SRC到44.1khz的,因此這會犯高通同樣的錯誤-雙重SRC(48khz->44.1khz->48khz),對音質的損害非常大。爲什麼不直接使用44.1khz的採樣率呢?
kernel
如我們所知,galaxy nexus用的是omap4460,音頻芯片是twl6040。因此我們下載omap的kernel代碼:
- $ git clone https://android.googlesource.com/kernel/omap.git
$ git clone https://android.googlesource.com/kernel/omap.git
詳見:http://source.android.com/source/downloading.html
就本篇的討論內容來看,我們只需關注如下幾個源文件:
1、sound\soc\codecs\twl6040.c
2、sound\soc\omap\omap-abe-dsp.c和sound\soc\omap\omap-abe.c
twl6040.c是音頻芯片twl6040的驅動代碼,這部分是通用的;
omap-abe是omap4460的音頻後端處理(Audio Back-End)驅動代碼,這是平臺相關的。結合後面的硬件框圖來看,就會明白audio_hw很大程度是直接控制abe。
其中omap的dsp代碼是以firmware的形式提供的,因此omap-abe-dsp.c用於調用dsp的接口函數。
hardware diagram
omap4460數據手冊及設計資料如下:
ABE:http://focus.ti.com/pdfs/wtbu/OMAP4430_ES2%20x_4460_ES1%200_PUBLIC_TRM_Addendum_ABE_HAL_vC.pdf
音頻系統框圖如下:
左邊是OMAP的ABE,右邊是codec twl4060,由此可知:音頻先經過OMAP ABE的處理,再送到codec輸出。
通話下行路徑:ABE[VX_DL -> DL_Mixer -> ...] -> PDM_DL -> CODEC[DAC -> Earpiece/Headfree/Headset]
其中在ABE端還要經過一些EQ、Gain、SRC、Echo部件,這裏不一一列出了。可見OMAP ABE是非常複雜的一個模塊,聲音在這裏處理好之後才送到CODEC,相比之下CODEC端的工作就簡單多了。